1. Field of the Invention
The present invention relates to the field of communication, and more particularly to a packet redundancy and recovery method for packet transmission.
2. Description of the Related Art
Real-time multi-media communication is typical UDP protocol based to avoid the un-expected timeout for TCP protocol. Typically, the packet loss is considered part of the channel characteristics and no try is made to recover the loss. A control channel may be used to feedback the packet loss information, so the sender will throttle the speed of sending to better match the effective end-to-end bandwidth. Some system may use multiple routes to send multiple copies of the data to achieve high quality. But nothing is done in a route to reduce the packet loss at the application level.
In the real-time audio and video communication over Internet, a lost packet represents only 30-60 milliseconds or so of length of time and is undetectable to our perception. A timeout for re-transmission will cause data to stop for 10 to 20 times of that length of time and is annoying. The current approach is to control the speed of sending to match the available bandwidth to reduce the amount of packet loss. For a network with high data loss rate, the effective bandwidth may become so low that real communication becomes impossible.
The traditional way to implement reliable transmission of data on an unreliable packet communication channel is to use acknowledgement, time out and re-transmission. Both ends have to wait for the time-out before a re-transmission can take place, and the time-out usually has to be longer than the round-trip time. Sufficiently long time out has to be used in a wide area network, and this has contributed to the low performance in using lossy network connections to transmit streaming data reliably.